Displaying 5 results from an estimated 5 matches for "silk24".
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silk
2014 Dec 11
2
PJSIP configuration question
Dan Cropp wrote:
> I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)?
>
> PJSIP is including the Contact for the ACK response to the OK.
> Contact:<sip:1234 at xxx.xxx.xx.xxx:5060>
>
There is no configuration option to configure this behavior. What is the
full SIP signaling?
--
Joshua
2014 Dec 11
0
PJSIP configuration question
...format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|opus|vp8|silk8|silk12|silk16|silk24), peer - audio=(gsm|ulaw|g729)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 66.241.99.161:11460
<--- SIP read from UDP:64.2.142.9...
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain
2016 Dec 10
6
failing to start asterisk on centos7
...egistered 'audio' codec 'silk' at sample rate '16000' with id '42'
== Created cached format with name 'silk16'
== Registered 'audio' codec 'silk' at sample rate '24000' with id '43'
== Created cached format with name 'silk24'
== Sorcery registered wizard 'bucket'
== Sorcery registered wizard 'bucket_file'
== Parsing '/etc/asterisk/sorcery.conf': Found
== Parsing '/etc/asterisk/stasis.conf': Found
== Parsing '/etc/asterisk/logger.conf': Found
== Message handler ...
2012 Aug 13
1
Websockets on Asterisk 11 and SipML5
Hello,
I'm trying to register a user using sipml5 on Asterisk 11. I followed the
instructions here:
http://thr3ads.net/asterisk-users/2012/08/1972342-Asterisk-Websockets
I added transport=ws to my sip.conf file:
[3002]
username=3002
secret=XXXXXXXXX
host=dynamic
type=friend
context=test
disallow=all
allow=g729
;allow=all ; Allow codecs in order of preference
allow=ilbc